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FFmpeg/libavcodec/mpc.c

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/*
* Musepack decoder
* Copyright (c) 2006 Konstantin Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file mpc.c Musepack decoder
* MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
* divided into 32 subbands.
*/
#include "avcodec.h"
#include "bitstream.h"
#include "dsputil.h"
#include "random.h"
#ifdef CONFIG_MPEGAUDIO_HP
#define USE_HIGHPRECISION
#endif
#include "mpegaudio.h"
#include "mpcdata.h"
#define BANDS 32
#define SAMPLES_PER_BAND 36
#define MPC_FRAME_SIZE (BANDS * SAMPLES_PER_BAND)
static VLC scfi_vlc, dscf_vlc, hdr_vlc, quant_vlc[MPC7_QUANT_VLC_TABLES][2];
static DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]);
typedef struct {
DSPContext dsp;
int IS, MSS, gapless;
int lastframelen, bands;
int oldDSCF[2][BANDS];
AVRandomState rnd;
int frames_to_skip;
/* for synthesis */
DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512*2]);
int synth_buf_offset[MPA_MAX_CHANNELS];
DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][36][SBLIMIT]);
} MPCContext;
/** Subband structure - hold all variables for each subband */
typedef struct {
int msf; ///< mid-stereo flag
int res[2];
int scfi[2];
int scf_idx[2][3];
int Q[2];
}Band;
static int mpc7_decode_init(AVCodecContext * avctx)
{
int i, j;
MPCContext *c = avctx->priv_data;
GetBitContext gb;
uint8_t buf[16];
static int vlc_inited = 0;
if(avctx->extradata_size < 16){
av_log(avctx, AV_LOG_ERROR, "Too small extradata size (%i)!\n", avctx->extradata_size);
return -1;
}
memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
av_init_random(0xDEADBEEF, &c->rnd);
dsputil_init(&c->dsp, avctx);
c->dsp.bswap_buf(buf, avctx->extradata, 4);
ff_mpa_synth_init(mpa_window);
init_get_bits(&gb, buf, 128);
c->IS = get_bits1(&gb);
c->MSS = get_bits1(&gb);
c->bands = get_bits(&gb, 6);
if(c->bands >= BANDS){
av_log(avctx, AV_LOG_ERROR, "Too many bands: %i\n", c->bands);
return -1;
}
skip_bits(&gb, 88);
c->gapless = get_bits1(&gb);
c->lastframelen = get_bits(&gb, 11);
av_log(avctx, AV_LOG_DEBUG, "IS: %d, MSS: %d, TG: %d, LFL: %d, bands: %d\n",
c->IS, c->MSS, c->gapless, c->lastframelen, c->bands);
c->frames_to_skip = 0;
if(vlc_inited) return 0;
av_log(avctx, AV_LOG_DEBUG, "Initing VLC\n");
if(init_vlc(&scfi_vlc, MPC7_SCFI_BITS, MPC7_SCFI_SIZE,
&mpc7_scfi[1], 2, 1,
&mpc7_scfi[0], 2, 1, INIT_VLC_USE_STATIC)){
av_log(avctx, AV_LOG_ERROR, "Cannot init SCFI VLC\n");
return -1;
}
if(init_vlc(&dscf_vlc, MPC7_DSCF_BITS, MPC7_DSCF_SIZE,
&mpc7_dscf[1], 2, 1,
&mpc7_dscf[0], 2, 1, INIT_VLC_USE_STATIC)){
av_log(avctx, AV_LOG_ERROR, "Cannot init DSCF VLC\n");
return -1;
}
if(init_vlc(&hdr_vlc, MPC7_HDR_BITS, MPC7_HDR_SIZE,
&mpc7_hdr[1], 2, 1,
&mpc7_hdr[0], 2, 1, INIT_VLC_USE_STATIC)){
av_log(avctx, AV_LOG_ERROR, "Cannot init HDR VLC\n");
return -1;
}
for(i = 0; i < MPC7_QUANT_VLC_TABLES; i++){
for(j = 0; j < 2; j++){
if(init_vlc(&quant_vlc[i][j], 9, mpc7_quant_vlc_sizes[i],
&mpc7_quant_vlc[i][j][1], 4, 2,
&mpc7_quant_vlc[i][j][0], 4, 2, INIT_VLC_USE_STATIC)){
av_log(avctx, AV_LOG_ERROR, "Cannot init QUANT VLC %i,%i\n",i,j);
return -1;
}
}
}
vlc_inited = 1;
return 0;
}
/**
* Process decoded Musepack data and produce PCM
* @todo make it available for MPC8 and MPC6
*/
static void mpc_synth(MPCContext *c, int16_t *out)
{
int dither_state = 0;
int i, ch;
OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE], *samples_ptr;
for(ch = 0; ch < 2; ch++){
samples_ptr = samples + ch;
for(i = 0; i < SAMPLES_PER_BAND; i++) {
ff_mpa_synth_filter(c->synth_buf[ch], &(c->synth_buf_offset[ch]),
mpa_window, &dither_state,
samples_ptr, 2,
c->sb_samples[ch][i]);
samples_ptr += 64;
}
}
for(i = 0; i < MPC_FRAME_SIZE*2; i++)
*out++=samples[i];
}
/**
* Fill samples for given subband
*/
static inline void idx_to_quant(MPCContext *c, GetBitContext *gb, int idx, int *dst)
{
int i, i1, t;
switch(idx){
case -1:
for(i = 0; i < SAMPLES_PER_BAND; i++){
*dst++ = (av_random(&c->rnd) & 0x3FC) - 510;
}
break;
case 1:
i1 = get_bits1(gb);
for(i = 0; i < SAMPLES_PER_BAND/3; i++){
t = get_vlc2(gb, quant_vlc[0][i1].table, 9, 2);
*dst++ = mpc_idx30[t];
*dst++ = mpc_idx31[t];
*dst++ = mpc_idx32[t];
}
break;
case 2:
i1 = get_bits1(gb);
for(i = 0; i < SAMPLES_PER_BAND/2; i++){
t = get_vlc2(gb, quant_vlc[1][i1].table, 9, 2);
*dst++ = mpc_idx50[t];
*dst++ = mpc_idx51[t];
}
break;
case 3: case 4: case 5: case 6: case 7:
i1 = get_bits1(gb);
for(i = 0; i < SAMPLES_PER_BAND; i++)
*dst++ = get_vlc2(gb, quant_vlc[idx-1][i1].table, 9, 2) - mpc7_quant_vlc_off[idx-1];
break;
case 8: case 9: case 10: case 11: case 12:
case 13: case 14: case 15: case 16: case 17:
t = (1 << (idx - 2)) - 1;
for(i = 0; i < SAMPLES_PER_BAND; i++)
*dst++ = get_bits(gb, idx - 1) - t;
break;
default: // case 0 and -2..-17
return;
}
}
static int mpc7_decode_frame(AVCodecContext * avctx,
void *data, int *data_size,
uint8_t * buf, int buf_size)
{
MPCContext *c = avctx->priv_data;
GetBitContext gb;
uint8_t *bits;
int i, j, ch, t;
int mb = -1;
Band bands[BANDS];
int Q[2][MPC_FRAME_SIZE];
int off;
float mul;
int bits_used, bits_avail;
memset(bands, 0, sizeof(bands));
if(buf_size <= 4){
av_log(avctx, AV_LOG_ERROR, "Too small buffer passed (%i bytes)\n", buf_size);
}
bits = av_malloc(((buf_size - 1) & ~3) + FF_INPUT_BUFFER_PADDING_SIZE);
c->dsp.bswap_buf(bits, buf + 4, (buf_size - 4) >> 2);
init_get_bits(&gb, bits, (buf_size - 4)* 8);
skip_bits(&gb, buf[0]);
/* read subband indexes */
for(i = 0; i <= c->bands; i++){
for(ch = 0; ch < 2; ch++){
if(i) t = get_vlc2(&gb, hdr_vlc.table, MPC7_HDR_BITS, 1) - 5;
if(!i || (t == 4)) bands[i].res[ch] = get_bits(&gb, 4);
else bands[i].res[ch] = bands[i-1].res[ch] + t;
}
if(bands[i].res[0] || bands[i].res[1]){
mb = i;
if(c->MSS) bands[i].msf = get_bits1(&gb);
}
}
/* get scale indexes coding method */
for(i = 0; i <= mb; i++)
for(ch = 0; ch < 2; ch++)
if(bands[i].res[ch]) bands[i].scfi[ch] = get_vlc2(&gb, scfi_vlc.table, MPC7_SCFI_BITS, 1);
/* get scale indexes */
for(i = 0; i <= mb; i++){
for(ch = 0; ch < 2; ch++){
if(bands[i].res[ch]){
bands[i].scf_idx[ch][2] = c->oldDSCF[ch][i];
t = get_vlc2(&gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7;
bands[i].scf_idx[ch][0] = (t == 8) ? get_bits(&gb, 6) : (bands[i].scf_idx[ch][2] + t);
switch(bands[i].scfi[ch]){
case 0:
t = get_vlc2(&gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7;
bands[i].scf_idx[ch][1] = (t == 8) ? get_bits(&gb, 6) : (bands[i].scf_idx[ch][0] + t);
t = get_vlc2(&gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7;
bands[i].scf_idx[ch][2] = (t == 8) ? get_bits(&gb, 6) : (bands[i].scf_idx[ch][1] + t);
break;
case 1:
t = get_vlc2(&gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7;
bands[i].scf_idx[ch][1] = (t == 8) ? get_bits(&gb, 6) : (bands[i].scf_idx[ch][0] + t);
bands[i].scf_idx[ch][2] = bands[i].scf_idx[ch][1];
break;
case 2:
bands[i].scf_idx[ch][1] = bands[i].scf_idx[ch][0];
t = get_vlc2(&gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7;
bands[i].scf_idx[ch][2] = (t == 8) ? get_bits(&gb, 6) : (bands[i].scf_idx[ch][1] + t);
break;
case 3:
bands[i].scf_idx[ch][2] = bands[i].scf_idx[ch][1] = bands[i].scf_idx[ch][0];
break;
}
c->oldDSCF[ch][i] = bands[i].scf_idx[ch][2];
}
}
}
/* get quantizers */
memset(Q, 0, sizeof(Q));
off = 0;
for(i = 0; i < BANDS; i++, off += SAMPLES_PER_BAND)
for(ch = 0; ch < 2; ch++)
idx_to_quant(c, &gb, bands[i].res[ch], Q[ch] + off);
/* dequantize */
memset(c->sb_samples, 0, sizeof(c->sb_samples));
off = 0;
for(i = 0; i <= mb; i++, off += SAMPLES_PER_BAND){
for(ch = 0; ch < 2; ch++){
if(bands[i].res[ch]){
j = 0;
mul = mpc_CC[bands[i].res[ch]] * mpc7_SCF[bands[i].scf_idx[ch][0]];
for(; j < 12; j++)
c->sb_samples[ch][j][i] = mul * Q[ch][j + off];
mul = mpc_CC[bands[i].res[ch]] * mpc7_SCF[bands[i].scf_idx[ch][1]];
for(; j < 24; j++)
c->sb_samples[ch][j][i] = mul * Q[ch][j + off];
mul = mpc_CC[bands[i].res[ch]] * mpc7_SCF[bands[i].scf_idx[ch][2]];
for(; j < 36; j++)
c->sb_samples[ch][j][i] = mul * Q[ch][j + off];
}
}
if(bands[i].msf){
int t1, t2;
for(j = 0; j < SAMPLES_PER_BAND; j++){
t1 = c->sb_samples[0][j][i];
t2 = c->sb_samples[1][j][i];
c->sb_samples[0][j][i] = t1 + t2;
c->sb_samples[1][j][i] = t1 - t2;
}
}
}
mpc_synth(c, data);
av_free(bits);
bits_used = get_bits_count(&gb);
bits_avail = (buf_size - 4) * 8;
if(!buf[1] && ((bits_avail < bits_used) || (bits_used + 32 <= bits_avail))){
av_log(NULL,0, "Error decoding frame: used %i of %i bits\n", bits_used, bits_avail);
return -1;
}
if(c->frames_to_skip){
c->frames_to_skip--;
*data_size = 0;
return buf_size;
}
*data_size = (buf[1] ? c->lastframelen : MPC_FRAME_SIZE) * 4;
return buf_size;
}
static void mpc7_decode_flush(AVCodecContext *avctx)
{
MPCContext *c = avctx->priv_data;
memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
c->frames_to_skip = 32;
}
AVCodec mpc7_decoder = {
"mpc sv7",
CODEC_TYPE_AUDIO,
CODEC_ID_MUSEPACK7,
sizeof(MPCContext),
mpc7_decode_init,
NULL,
NULL,
mpc7_decode_frame,
.flush = mpc7_decode_flush,
};